The Internet telephony or voice over Internet protocol (VoIP) is defined as communication technology that converts voice data into Internet protocol (IP) data packets, which can be transmitted through communication networks, in order to support voice conversation services not only through a telephone network but also over the Internet.
A major advantage of the VoIP and Internet technology is that they provide a telephone service by utilizing the existing IP networks in an untouched state, such that telephone users can be provided with long distance and international telephone services in Internet and intranet environments while they pay local call rate.
The VoIP was introduced by major equipment providers, such as Cisco, VocalTec, 3Com and NetSpeak, in an attempt to promote the use of ITU-T, H.323 and the like, which are standards for transmitting voice or sound using IP over the public Internet or through an intranet of a company. In order to promote the directory service standard, the VoIP forum allows users to locate other users. Furthermore, automatic call distribution and the use of touch-phone signals for voice mails are also enabled.
As a characteristic feature, the VoIP uses real time protocol (RTP) in order to support on-time arrival of packets in addition to its original IP function. If a common public network is used, the characteristics of best-effort services make it difficult to support quality of service (QoS) for voice communication. As a result, the VoIP services can be provided with higher quality when a private network managed by a separate enterprise or an Internet telephone service provider (ITSP) is used.
The application layer gateway (ALG) is provided by an application gateway or a router. According to the ALG, a conventional firewall or network address translator (NAT) can use several protocols in order to inspect packets transmitted between internal and external networks, and a verification process can be performed such that dynamically-allocated network resources can pass through the firewall or NAT.
That is, the ALG is a device that routes a message packet, dynamically entering from an external network resource, to a specific host of a security-maintained internal network through inspection and verification procedures.
In particular, the session initiation protocol-ALG (SIP-ALG) is technology designed to support communication between a SIP signaling gateway or a media gateway in an internal private network and a SIP proxy server or a SIP agent in an external public network.
FIG. 1 is an overview of the construction of a VoIP network that provides typical Internet telephony services.
As shown in FIG. 1, a private VoIP network using VoIP services generally includes an SIP signaling gateway 1, a media gateway 2, an internal SIP terminal 3 and an SIP-ALG router 4.
The SIP-ALG router 4 of the private VoIP network is connected to an external SIP proxy server 6 and external SIP terminals 7 and 8 through the Internet 5.
The SIP-ALG router 4 binds a SIP signaling message to a VoIP RTP stream that the SIP signaling gateway 1 or the media gateway 2 transmits. The SIP-ALG router 4 parses the bound packet, and then substitutes the private IP address of the packet with its own public IP address before routing the packet over the Internet.
Based on the ALG technology as described above, the SIP signaling gateway 1 and the media gateway 2 of the private VoIP network can communicate with the SIP proxy server 6 in the external network, which uses a public IP address, by transmitting/receiving packets to/from the SIP proxy server 6.
However, as shown in FIG. 1, a typical private VoIP network is constructed so that the SIP-ALG router 4, supporting the SIP-ALG function, is separated from the SIP signaling gateway 1 and the media gateway 2.
In this construction separated from the SIP signaling gateway 1 and the media gateway 2, the SIP-ALG router 4 overcomes the following problems of the private VoIP network.
When any one of the SIP agents 1 to 3 in the private VoIP network attempts to set up a session with an external user, following problems may occur. Firstly, the private IP address of the SIP agent 1 or 2, described in a SIP message or a Session Description Protocol (SDP), cannot be routed from outside. As a second problem, the NAT or the firewall does not allow traffics from outside to pass through unless they are permitted. Finally, traffic cannot be transmitted from inside to outside unless it is allowed by the firewall.
Due to these problems, the SIP agents 1 and 2 and the SIP terminal 3 inside the firewall cannot communicate with the external SIP proxy server 6 or the external SIP terminal 7 or 8.
Accordingly, the SIP-ALG router 4 is designed to solve the NAT- and firewall-related problems and to convert the private IP address into public IP address.
However, the SIP signaling gateway 1, the medial gateway 2 and the router 4 supporting the SIP-ALG functions are required to be separately purchased in order to operate the foregoing private VoIP network. Of course, a company or firm operating the private VoIP network has to separately maintain, repair and manage respective pieces of equipment.
Accordingly, it is difficult for a network operator, who has to manage several pieces of equipment, to easily manage the private VoIP network.
In the meantime, an integrated Internet telephony system having an all-in-one architecture in which SIP signaling, media gateway and SIP-ALG functions are integrated can be provided by realizing the SIP signaling gateway 1, the media gateway 2 and the SIP-ALG router 4 in one piece of equipment in order to facilitate the maintenance, repair and management of the equipment while minimizing construction costs of the private VoIP network.
Also, there is a problem in that the SIP signaling gateway 1, the media gateway 2 and the SIP-ALG router 4 are required to be allocated with different unique IP addresses while the integrated Internet telephony system uses only one IP address.